304 results about "Speech rate" patented technology

The invention relates to a method for automatic evaluation based on generalized fluent spoken language fluency, which comprises the following steps of: acquiring speech data according to different ages and spoken language levels by using a speech input device; adopting an evaluating model based on characteristics of the generalized fluency and the machine learning training fluency; configuring a speech recognition system with corresponding parameters according to scripts of different subjects and genders of enunciators in the speech data; performing quantification on speech speed coherence, content understanding, advanced skills and reconstruction standard characteristics in the speech data to comprehensively extract the characteristics of the fluency from the speech data from the angle of expert assessment and evaluation; and adopting a decision tree method in regression fitting analysis and data mining to detect faults of abnormal fluency and grade and diagnose the fluency. The acquired score of the machine fluency can reach the level close to that of grading experts, and the relativity index exceeds that of 2 to 3 of general 5 experts; besides, the method has a high speed, and can be embedded into a spoken language automatic evaluation system to serve as an important module to evaluate fluency indexes in pronunciation quality.

The present invention provides a digital voice book library system which through Internet provides an customized digital audio version book that can be customized according to a learner's learning language, learning level, learning condition, learning taste or the like by a learner's selecting/varying audio attributes of the audio version book(digital audio file) associated with the equivalent printed script (hardcopy) version book that is owned by the learner or is borrowed from a conventional library or a book rental store. Wherein the audio attribute of said customized digital audio version book can be varied and customized in language, voice, speech speed or playback device at every repetition time according to user's learning language, learning level, learning condition or the like, corresponding to the equivalent script book despite different audio attribute. Using the present invention a learner can repeatedly listen to a same content audio version book without boring, therefore can acquire a good language ability in short time.

The invention discloses an automatic voice quality inspection method, a system thereof, a device and a storage medium. The method includes the following steps: obtaining a voice file to be inspected;processing the voice file to be inspected, and converting the voice file to be inspected into a structured text file; scoring the voice file to be inspected and the structured text file according to apreset quality inspection scoring dimension and a preset score rule, and obtaining the score corresponding to each quality inspection scoring dimension; wherein, the quality inspection scoring dimension includes speech rate and silent analysis scoring, emotional analysis scoring, keyword and sensitive word detection scoring, and quality inspection rule matching scoring; and obtaining the final score of the voice file to be inspected according to the score corresponding to each quality inspection scoring dimension. The method can realize automatic voice quality inspection, and improves qualityinspection efficiency and accuracy.

A method for varying speech speed is provided. The method includes the following steps: receive an original speech signal; calculate a pitch period of the original speech signal; define search ranges according to the pitch period; find a maximum within each of the search ranges of the original speech signal; divide the original speech signal into speech sections according to the maxima; obtain a speed-varied speech signal by applying a speed-varying algorithm to each speech section of the original speed signal according to a speed-varying command; and eventually, output the speed-varied speech signal.

An electronic device for adjusting a speech output rate (speech rate) of speech output data.

Improved methods of presenting speech prompts to a user as part of an automated system that employs speech recognition or other voice input are described. The invention improves the user interface by providing in combination with at least one user prompt seeking a voice response, an enhanced user keyword prompt intended to facilitate the user selecting a keyword to speak in response to the user prompt. The enhanced keyword prompts may be the same words as those a user can speak as a reply to the user prompt but presented using a different audio presentation method, e.g., speech rate, audio level, or speaker voice, than used for the user prompt. In some cases, the user keyword prompts are different words from the expected user response keywords, or portions of words, e.g., truncated versions of keywords.

The invention discloses an automatic speech rate adjusting method. The method comprises the steps that input speech information is acquired; speech feature information of the speech information is extracted; the speech information playing rate corresponding to the speech feature information is inquired from a speech database; the speech information playing rate is adjusted according to the playing rate. According to the method, the preset playing rate corresponding to the speech feature information can be determined according to the speech feature information of the speech information input in real time, and the speech rate of the input speech information is adjusted according to the playing rate to meet the requirements various users; not only is the purpose of self-adaptively adjusting the playing rate according to the content of the speech information achieved, but also the adaptability is high when the method is applied to the occasions of communication, program playing and the like. The invention further discloses a terminal which can achieve the purpose of self-adaptively adjusting the playing rate according to the content of the speech information.

An electronic device and a method are provided. The electronic device includes an audio input module configured to receive a speech of a user as a voice input, an audio output module configured to output content corresponding to the voice input, and a processor configured to determine an output scheme of the content based on at least one of a speech rate of the speech, a volume of the speech, and a keyword included in the speech, which is obtained from an analysis of the voice input.

The invention relates to a method for improving speech recognition rate with gain self-adapting. The invention is characterized in that: by evaluating noise, adjusting recording gain and adjusting port detection parameters to improve speech rate. The inventive method comprises steps of: 1, evaluating background noise; 2, adjusting recording gain according to the evaluated background noise in step 1; taking port detection and speech recognition on the basis of step 1 and step 2.

The invention discloses a system for realizing sound speed variation and tone variation, which comprises an input cache module, a tone variation processing module, a speed-variation no-tone-variation processing module and a data output module, wherein the input cache module is used for reading the sound signal data to be processed into the cache; the tone variation processing module is used for carrying out the tone variation processing on the sound signal to change the sound tone; the speed-variation no-tone-variation processing module is used for carrying out the speed-variation no-tone-variation processing on the sound signal, thereby changing the sound speed without changing the tone; and the data output module is used for outputting the speed-variation tone-variation signal. The speed-variation no-tone-variation processing module comprises a segmentation data module and a connection data module, wherein the speed-variation no-tone-variation processing module extracts a string of signal subfamilies (namely small sections of sound) from the original speech signal according to the coefficient of variation in speed by using a window function; and the connection data module connects the signal subfamilies according to the time sequence, thereby obtaining the speed-variation no-tone-variation signal. The invention realizes the speed-variation no-tone-variation function and the speed-variation tone-variation function of the audio frequency by using very low algorithm complexity, and does not introduce noise, thereby enhancing the quality of the processed sound.

A speaking-rate dependent prosodic model builder and a related method are disclosed. The proposed builder includes a first input terminal for receiving a first information of a first language spoken by a first speaker, a second input terminal for receiving a second information of a second language spoken by a second speaker and a functional information unit having a function, wherein the function includes a first plurality of parameters simultaneously relevant to the first language and the second language or a plurality of sub-parameters in a second plurality of parameters relevant to the second language alone, and the functional information unit under a maximum a posteriori condition and based on the first information, the second information and the first plurality of parameters or the plurality of sub-parameters produces speaking-rate dependent reference information and constructs a speaking-rate dependent prosodic model of the second language.

The present invention relates to a method and apparatus for improving spontaneous speech recognition performance. The present invention is directed to providing a method and apparatus for improving spontaneous speech recognition performance by extracting a phase feature as well as a magnitude feature of a voice signal transformed to the frequency domain, detecting a syllabic nucleus on the basis of a deep neural network using a multi-frame output, determining a speaking rate by dividing the number of syllabic nuclei by a voice section interval detected by a voice detector, calculating a length variation or an overlap factor according to the speaking rate, and performing cepstrum length normalization or time scale modification with a voice length appropriate for an acoustic model.

The invention discloses a voice recognition method and device. The method comprises the following steps: acquiring voice information input by a user; recognizing entity information in the voice information based on a named entity recognition system; acquiring speed information and volume information of the user when the voice information is input; and recognizing emotive information corresponding to the voice information according to the entity information, the speed information and the volume information based on a deep learning emotion analysis model. According to the voice recognition method in the embodiment of the invention, the voice information input by the user is acquired, the entity information in the voice information is recognized based on the named entity recognition system, the speed information and the volume information of the user when the voice information is input are acquired, and the emotive information corresponding to the voice information is recognized according to the entity information, the speed information and the volume information based on the deep learning emotion analysis model and is taken as an important factor for voice recognition, so that the accuracy rate of the voice recognition is increased, and the real demands of a user can be well met.

The invention discloses a text speech synthesis method after speaker emotion simulated optimization translation. Voice information of a user is obtained; The background analyzes an audio file to obtain frequency and speech speed parameters; the background is introduced to a voiceprint identification system to obtain gender and age parameters; speech is recognized to obtain text information; an emotion parameter is obtained from the text via analysis on grammar, vocabulary and sentences of the text; frequency, speech speed, gender, age and emotion features are combined, and a characteristic value is set for each feature; and the characteristic values are combined with a speech synthesis SSML grammar to set the broadcast speed, volume and word pause in the speech synthesis SSML grammar. Thus, synthesized speech broadcast of another language can reflect the emotion feature of a native language of a speaker. The mood, tone, vocabulary and grammar features of the speaker are identified, sothat the speech translated synthesis broadcast reflect the emotion of the speaker at present.